VoIP (Voice over Internet Protocol) is one of the popular communication technology. In VoIP, SIP (Session Initiation Protocol) defined by IETF is the most widely used protocol because of its simple structure, expandability and easy operation.
In the present SIP Internet environment, more and more users install NAT (Network Address Translator) servers, but NAT servers induce the communication failure for RTP voice packets after SIP messages.
Referring to FIG. 1, which shows the SIP (Session Initiation Protocol) network environment for VoIP, comprises NAT server 1, NAT server 2 and SIP proxy server 3. SIP proxy server 3 is responsible for conducting SIP, i.e. for registration, forwarding or redirection of the Internet extension 2178 and Internet extension 2167 (SIP client's terminals).
Internet extension 2178 (IP: 192.168.1.2) and Internet extension 2167 (IP: 192.168.1.3) are under Taiwan NAT server 1 (IP: 140.124.40.11) and USA NAT server 2 (IP: 163.21.34.55) respectively, voice packets must be transferred through RTP-relay in SIP proxy server 3, client to client (C2C) communication between Internet extension 2178 and Internet extension 2167 is impossible. When a plurality of client's terminals communicates through SIP proxy server 3 simultaneously, it is apparent that the communication efficiency will be reduced significantly.
Referring to FIG. 2, the communication between Taiwan Internet extension 2178 and USA Internet extension 2167 is described. Taiwan Internet extension 2178 firstly issues INVITE 2167 message, in which SDP (192.168.1.2:6000) represents the IP address of the Internet extension 2178 (please see “Introduction of the Session Initiation Protocol (SIP)” on page 9). INVITE 2167 message pass through Taiwan NAT server 1, SIP proxy server 3, USA NAT server 2 and reach USA Internet extension 2167, causing USA Internet extension 2167 issues 180 Ringing message to feed back; when USA Internet extension 2167 picks up the phone, a 200 OK message is generated, in which SDP (192.168.1.3:1200) represents the IP address of the USA Internet extension 2167 (please see “Introduction of the Session Initiation Protocol (SIP)” on page 9), and then feed back. According to SIP communication protocol, Internet extension 2178 and Internet extension 2167 will then transfer voice packets each other for communication, but because the IP addresses 192.168.1.2 and 192.168.1.3 of the Internet extension 2178 and Internet extension 2167 are virtual addresses, so actually the voice packets cannot transfer each other for communication.
There is an improvement as shown in FIG. 3. An RTP-Relay (Real Time Transport Protocol-Relay) 4 is added on the SIP proxy server 3, such that the SIP proxy server 3 can change the message SDP (192.168.1.2:6000) into SDP (202.145.2.1:1200), in which “202.145.2.1” is the actual IP address of RTP-Relay 4. Similarly, when USA Internet extension 2167 issues 200 OK message, the SIP proxy server 3 can change the message SDP(192.168.1.3:1200) into SDP(202.145.2.1:1201), in which “202.145.2.1” is the actual IP address of RTP-Relay 4. Since “202.145.2.1” is the actual IP address of RTP-Relay 4, the voice packets of the Internet extension 2178 and Internet extension 2167 can transfer each other through the actual IP address “202.145.2.1” for communication.
However, there is a disadvantage in RTP-Relay 4, the bandwidth for video communication of RTP-Relay 4 is 2 Mb/sec, so the expense for one month per user is NT$20,000. If there are 1 million users to communicate simultaneously, the expense of bandwidth for RTP-Relay 4 will be NT$20 billion/month, therefore this method is not useful.